1. Field of the Invention
The field of the invention is that of the compression of digital audio signals. The invention can be applied notably to the transmission of sound signals on digital channels as well as to devices for the storage of digital sound signals.
More precisely, the invention concerns a bit allocation device enabling an adaptive quantization of a digital audio signal, after this signal has been transformed into the frequency domain and cut up into frequency bands.
The invention may be implemented, for example, in direct satellite broadcasing systems such as those developed in the European DAB (Digital Audio Broadcasting) project, or again in ISDN broadcasting and high fidelity distribution systems. It can also be applied notably to storage devices such as digital disks.
Digital audio signals have many advantages over analog signals, notably as regards the fidelity of the sound, the preservation of the initial quality and flexibility of use. However, the bit rate resulting from the conversion of the audio signals into digital form is very high, especially for high quality signals, the bandwidth of which is greater than 15 kHz.
It is then necessary to use bit rate reduction techniques.
2. Description of the Prior Art
In a known and widespread way, the techniques used employ algorithms for the mathematical transformation of the source digital audio signal. The transform coding techniques have been extensively applied to the fields of images or of speech. Since very recently, they are also applied to the processing of audio (chiefly musical) signals.
In existing coders implementing these techniques, the signal is first of all cut up into temporal blocks, and is then subjected to a time/frequency transformation. It is the coefficients of the transformed blocks that are encoded and transmitted. At the decoder, a reverse transformation delivers the decoded and reconstructed signal.
The application of mathematical transformation achieves a concentration of the energy of the source signal on the biggest coefficients, and thus enables a reduction of the bit rate by controlling the auditive degradation and reducing it to the minimum, notably by the selective elimination of certain of the transformed coefficients. Indeed, the fact of working in the frequency domain contributes towards taking account of perceptual and psychoauditive properties that are mainly linked to the spectral nature of the sound. The taking into account of the psychoauditive criteria in most existing devices is based on the analysis by ZWICKER in Psychoacoustique (Psychoacoustics), Masson, 1981, based on the concept of the masking of inaudible spectral components.
The known devices made on the basis of these principles differ from one another in certain preferences as regards their designing:
the transmission or non-transmission of a piece of auxiliary information (side information) to the main information;
the use or non-use of techniques overcoming the effect of transmission disturbances;
the techniques of taking account of the psychoauditive criteria to achieve the bit rate reduction and the localization of their implementation in the signal coding and decoding chains;
Thus, in a first known device of this type, as described in the French patent document No. 89 06194, "Procede et installation a codage des signaux sonores" (Sound Signal Coding Method and Equipment) filed on behalf of the present Applicants, the following are implemented successively: the cutting up of the sound signal into blocks of samples, the time/frequency transformation and a predictive and adaptive coding of the most significant coefficients of each block, using a stationarity of the signal. In this device, the auxiliary information is transmitted during transition blocks, thus making it impossible to take account of an inter-block correlation. In all the other situations, this auxiliary information is used solely to control the bit allocation module supplying the main signal quantizer. This device enables a reduction in the bit rate. However, it leads to a chain degradation of the reconstitution of the blocks received when an error occurs, because this error gets passed on to the next block, and so on and so forth, through the loop for preparing the auxiliary information controlling the bit allocator and the quantizer of the decoder.
There are also known devices in which a piece of auxiliary information is transmitted for each block, by adaptive coding. Such a device is described, for example, in the article by Bochow, "Multiprocessor Implementation of an ATC Audio Codec" (Acts of the ICASSP Congress 1989, Glasgow). A drawback of this device is that the continuous coding of the auxiliary information calls for a high bit rate, to the detriment of the bit rate allocated to the main information.
The document by Johnston, "Transform Coding of Audio Signals Using Perceptual Noise Criteria", IEEE Journal on Selected Areas in Communication", Vol. 6, No. 2, February 1988, pp. 314-323, has a bit reduction device using adaptive quantization, with application of the masking thresholds according to Zwicker's analysis in the form of a prediction algorithm applied at the quantizer of the main signal. This algorithm seeks to minimize the noise-to-masking threshold ratio. Just as in Bochow's device, the auxiliary information is transmitted continuously. Furthermore, this device has a variable length coding or Huffman coding at output of the quantizer, which is quite complicated to apply.
The invention is aimed at overcoming the drawbacks of these different known devices.